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[Curtis H List (Too Tall)]

Montez:

With the information provided we can only make guesses at the processing that might be appropriate. Adding the screenshots Timo asked for would certainly help. The measurements should be on-axis with both magnitude and phase plots, though an average of where a musician would stand - left / center / right of the wedge - would also be worth a look.

Even that will lead us to making only somewhat better guesses. The best thing would be to send the wedge around the world to each of us that would like to give you input on the processing so we could do the measurements and tuning using live measurement data. Prepaid of course.

But what if you could effectively do that, for free, without sending the wedge anywhere?

You can. It's easy. It's what I have been doing for several years. It allows you to make the measurement once and simply replay it anytime you want to change something.

This Friday I'm going to tune a pair of Danley SH100's and a TH112 that my employees hung in a small auditorium today. Yesterday I made the measurements* while the loudspeakers were at my shop. Tomorrow morning I'm going to do the processing in the electrical domain with a loop through of the measurement recordings through a processed Ashly amplifier. Friday will see a few parametric input filters added for room specific issues.

Anyone that would like to play along can literally do so with a dual channel FFT measurement system and a loudspeaker processor. The 10 second 48k, 16-bit stereo recordings have the loudspeaker measurement on the left channel and the pink noise reference on the right channel. It's helpful if your measurement software can playback a user selected file for its stimulus output. Simply do a loopback of the right output of your audio hardware to the right input. Route the audio hardware's left output to the processor's input and route the processor's output to the audio hardware's left input. Tune away. Both SH100 and TH112 were exactly 26' away from the microphone and the measurement mic gain was identical for both. Thus you can not only EQ the loudspeakers as you'd like, but add crossovers, perform the time/phase alignment (assume for now that the cabinets will be equidistant to the listener) and set the relative levels of the two passbands.

SH100 Measurement Recording

TH112 Measurement Recording

===

In your case Montez, you'd want to make the same kind of stereo recording with the mic at ear position like this:



A smooth cement surface is ideal, but even a 4x8 sheet of plywood in the yard with the wedge in the middle will do nicely in this situation. Low wind is very helpful for stable phase traces at HF, but that's not critical in this application either. You'll need enough level from the pink noise stimulus into the wedge to achieve good coherence and S/N through the range of interest with your two passbands (at least a couple of octaves on either side of the expected crossover point). You should be able to significantly outperform this since the mic will be so close to the source.

One last thought, it's not polite to send full range pink noise through a HF compression driver, so I've supplied a link for you to download a 20 second sample that has been high passed with a 4th order Butterworth at 200Hz. Just use your software's pink noise generator for the woofer.

Pink Noise with 200Hz HP

===

* Before long I'm going to walk through some of the basics of making good measurements. It'll be reflection-free and ground plane based, i.e. ideal, for reasons that'll be covered as well. It's not as easy as it sounds to do correctly, but it's entirely possible depending on the weather and the cops.

Finally, I appreciate and agree with Paul's terse reply concerning the knowledge required to do this correctly. It requires a huge amount of empirical study (time, effort, money) to begin to learn how much you don't know. You're starting well, attend Smaart and Syn-Aud-Con seminars. Syn-Aud-Con has also added some extremely well done online seminars that are much less expensive than live training. Blow things up and post questions in the mean time.

Hey Langston,
Are you still a member of SAC?

Remember the room convolution and de-convolution discussions?
At the beginning they used Praxis in the classes.
http://www.libinst.com/Room%20Sound%20Convolution.pdf

If you have the Liberty Instruments PRAXIS AudPod hardware, you can easily measure the
binaural IR of the room, letting the measurement system handle both the stimulus (the signal
that drives the sound system) and the acquisition (the signal picked up by the microphones, in
this case).

This is fast, but it does require you to run cables from the AudPod to both the sound
system s panel and from your measurement microphones, which may be inconvenient and prone
to hum pickup.
If you don t have the PRAXIS AudPod, or if you want to avoid running long cables, you can
instead play an audio CD to provide the stimulus sound, as will be described below. You can
then record the result using a DAT recorder, or recording software on a laptop computer with a
good quality soundcard. The resulting file can then be processed using the PRAXIS software to
produce the needed binaural IR file. This processing can be done using PRAXIS in its (free)
demo mode.

You can play a CD with the test signal on it into the speaker and record the test signal and what the microphone hears.
Note that the test signal is on this website free to download.

Next you use the free demo of Praxis to de-convolute the recording to extract the IR.
In the end you want only one channel with the recording on it.
This can be done post process in Praxis.

Something to look at if you do not have Smaart and other ways to de-covolute the signal involved.
NOTE: One of the companies more connected to SAC made a free of charge program to deal with de-convolution and convolution.

[Stu McDoniel]

 What mics are you using?   Go with Audix OM7's or EV ND 967's
You have a 15" driver and you really dont want to go much higher then you already are with you xover choice.  24db per octave
linkwitz-riley good!   If you have to notch it as far as you do in the regions you mentioned something is amiss.  Start from scratch
and flatten out the EQ and run pink noise and ajust the the compression driver/horn -loudspeaker levels so it is as flat as possible
without applying and GEQ or PEQ.   Do not start by setting it by ear.  I think  that was your first mistake and and snowball starting rolling
and getting bigger and that was setting the level between your drivers by ear then running pink noise.    Also, most compression drivers
are pretty dam hot at 2.5k to the human ear and this is where PEQ with a real narrow slice can grab that "bite your head off" area
and notch it out and that can be anywhere in the 2k-3.5k regions.   If you are running rappers then by all means dump the 58's
and go with the  Audix OM7 Or EV 967.
 

[Jim McKeveny]

Your speaker equipment is baseline reasonable. The PGX transmitters, however, have horrible response-altering companding, and perhaps the TX gain is too high also. Try a wired 58 side by side with the PGX, and see how your box sounds.

[Jim McKeveny]

Another thing: are you using any CD Horn EQ? While the B&C claims Constant Directivity, the bell itself does not show the required pinched throat/diffraction slit.

[Montez Carter]

Another thing: are you using any CD Horn EQ? While the B&C claims Constant Directivity, the bell itself does not show the required pinched throat/diffraction slit.

No I am not using any CD horn EQ I spent some time on these with pink noise getting them flat according to my smaart measurements. Which to my ear don't sound near as bright but seams to have more mid range vocal clarity. I also changed out my PGX series with 58 capsules for the Line 6 70 and I am getting a lot more gain before feedback without almost any graphic and the mysterious 800-1.2k problem went away.  Curious as that was around my low to high pass freq. But when I measured the cab my comp driver had to be lowered almost -10db. They work now but hell I have spent the last 6 weeks watching you tube and figuring out this Smaart program.

[Tim McCulloch]

No I am not using any CD horn EQ I spent some time on these with pink noise getting them flat according to my smaart measurements. Which to my ear don't sound near as bright but seams to have more mid range vocal clarity. I also changed out my PGX series with 58 capsules for the Line 6 70 and I am getting a lot more gain before feedback without almost any graphic and the mysterious 800-1.2k problem went away.  Curious as that was around my low to high pass freq. But when I measured the cab my comp driver had to be lowered almost -10db. They work now but hell I have spent the last 6 weeks watching you tube and figuring out this Smaart program.

Can you post a screen shot of the magnitude, phase and coherency traces?

[Jim McKeveny]

 But when I measured the cab my comp driver had to be lowered almost -10db.
[/quote]


-10db off the previous -10db you indicated? That seems extreme.

[Montez Carter]

But when I measured the cab my comp driver had to be lowered almost -10db.



-10db off the previous -10db you indicated? That seems extreme.

Yes -10db from my previous setting. I got into chasing my tail on these and had them way to bright. I also took my initial measurements with a behringer measurement mic which I changed to a Beyerdynamic a friend let me borrow for my recent measurements. I knew it sounded bright to me thats why I should have measured before I set the DSP in the amplifiers. Yes I know that is extreme. I knew I had too much comp driver when I heard a friends EAW SM15s at war volume and no feedback and I noticed hmmm not as bright as my cabs and the light bulb came on.

[Jeff Babcock]

I am just using the DSP from the crowns(crossover and gain only) and I have not did any parametric cuts as I did them all with the 31 band.



I have spent the last 6 weeks watching you tube and figuring out this Smaart program.

Your 31 band EQ should be for dealing with issues at the gig, not for speaker tuning.  That's like using a chainsaw to do brain surgery.

Don't underestimate how complex it is to design a good preset.  I am comfortable with Smaart and measurements after many years of doing them, but the more I learn, the more I realize that I didn't know about/don't fully understand.  If you are just starting into this, be prepared to have your eyes opened regarding how many variables that there are to consider - for instance, right now you seem to only be thinking about magnitude, but not phase....

While you are gaining knowledge, ship one of your speakers to Langston or another measurement guru and pay them to make you a preset.  You'll not only get a great result, but you will learn a lot in the process.  It will be well worth the expense and be far better than what you'll be able to achieve on your own at this point.  It takes a lot longer than 6 weeks of watching Youtube videos/ learning Smaart to be "good" at this, let alone "great".

Best regards
Jeff

[Tim McCulloch]

Your 31 band EQ should be for dealing with issues at the gig, not for speaker tuning.  That's like using a chainsaw to do brain surgery.

Don't underestimate how complex it is to design a good preset.  I am comfortable with Smaart and measurements after many years of doing them, but the more I learn, the more I realize that I didn't know about/don't fully understand.  If you are just starting into this, be prepared to have your eyes opened regarding how many variables that there are to consider - for instance, right now you seem to only be thinking about magnitude, but not phase....

While you are gaining knowledge, ship one of your speakers to Langston or another measurement guru and pay them to make you a preset.  You'll not only get a great result, but you will learn a lot in the process.  It will be well worth the expense and be far better than what you'll be able to achieve on your own at this point.  It takes a lot longer than 6 weeks of watching Youtube videos/ learning Smaart to be "good" at this, let alone "great".

Best regards
Jeff

Yes, yes, yes.  I made hundreds of SpectraLab/SMAART measurements that turned out to be invalid for the purpose at hand.  After I was able to make consistently valid measurements I had to learn how to translate the display into electronic actions that gave acoustic results appropriate for the purpose at hand.

Then I had to re-learn everything I thought I knew about tuning speaker crossovers.  I'm still learning, which is why I read everything Ivan, Tom, Dave G or Langston post.

Montez, don't let us discourage you; this exercise will affect your thinking for years to come, and in a good way.

[ProSoundWeb Community]